Audio signal processing has advanced in many ways and becomes increasingly important. In audio signal processing, Low-Delay Unified Speech and Audio Coding aims to provide coding techniques suitable for speech, audio and any mixture of speech and audio. Moreover, LD-USAC aims to assure a high quality for the encoded audio signals. Compared to USAC (Unified Speech and Audio Coding), the delay in LD-USAC is reduced.
When encoding audio data, a LD-USAC encoder examines the audio signal to be encoded. The LD-USAC encoder encodes the audio signal by encoding linear predictive filter coefficients of a prediction filter. Depending on the audio data that is to be encoded by a particular audio frame, the LD-USAC encoder decides, whether ACELP (Advanced Code Excited Linear Prediction) is used for encoding, or whether the audio data is to be encoded using TCX (Transform Coded Excitation). While ACELP uses LP filter coefficients (linear predictive filter coefficients), adaptive codebook indices and algebraic codebook indices and adaptive and algebraic codebook gains, TCX uses LP filter coefficients, energy parameters and quantization indices relating to a Modified Discrete Cosine Transform (MDCT).
On the decoder side, the LD-USAC decoder determines whether ACELP or TCX has been employed to encode the audio data of a current audio signal frame. The decoder then decodes the audio signal frame accordingly.
From time to time, data transmission fails. For example, an audio signal frame transmitted by a sender is arriving with errors at a receiver or does not arrive at all or the frame is late.
In these cases, error concealment may become useful for ensuring that the missing or erroneous audio data can be replaced. This is particularly true for applications having real-time requirements, as requesting a retransmission of the erroneous or the missing frame might infringe low-delay requirements.
However, existing concealment techniques used for other audio applications often create artificial sound caused by synthetic artefacts.